Re: [Serusers] RTPproxy debug

I dont know If the problem is realy of my ser.cfg, because the IP phones with public IPs work, but the phones with private IPs just ring... but I believe theres no RTP passing. Theres my SER: |--------------| | |-------------- private IP | S E R | | |-------------- public IP |--------------| Theres my ser.cfg: (...) listen = 193.136.xxx.xxx listen = 10.0.0.135 (...) alias=fccn.pt alias=193.136.xxx.xxx alias=10.0.0.135 # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=1 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/postgres.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" #loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/options.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" #loadmodule "/usr/local/lib/ser/modules/xdz_tools.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- nathelper params -- # proxy nat modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) (...) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # -------------------------------------------------- # NOTIFY Keep-Alive Section # -------------------------------------------------- if ((method=="NOTIFY") && search("^Event: keep-alive")) { sl_send_reply("200", "OK"); break; }; if ((method=="NOTIFY") && (uri=~"^sip:700@")) { sl_send_reply("200", "OK"); break; }; # -------------------------------------------------- # OPTIONS Section # -------------------------------------------------- if (method=="OPTIONS") { options_reply(); break; }; if (method=="REGISTER" && nat_uac_test("3")) { fix_nated_contact(); force_rport(); setflag(2); }; # -------------------------------------------------- # Registration Section # -------------------------------------------------- (...) # -------------------------------------------------- # Accounting Section # NOTE: We test for flag 14 because we do not want to record Click2Dial # entries # -------------------------------------------------- if ((method=="INVITE" || method=="BYE" || method=="CANCEL"||method=="ACK" ||method=="REGISTER") && !isflagset(14)) { setflag(1); }; # -------------------------------------------------- # NAT Tear-Down Section # -------------------------------------------------- if ((method == "BYE" || method == "CANCEL")) { unforce_rtp_proxy(); }; # -------------------------------------------------- # Record Route Section # -------------------------------------------------- # we record-route all messages -- to make sure that subsequent messages # will go through our proxy; that's particularly good if upstream and # downstream entities use different transport protocol if (!method=="REGISTER") { record_route(); }; # -------------------------------------------------- # Loose Route Section # # Grant route routing if route headers present # -------------------------------------------------- if (loose_route()) { route(2); break; }; # # -------------------------------------------------- # NAT Test Section #1 # -------------------------------------------------- if (nat_uac_test("3") && !search("^Record-Route:")) { force_rport(); fix_nated_contact(); }; (...) # -------------------------------------------------- # Alias Routing Section # -------------------------------------------------- lookup("aliases"); if (!uri==myself) { route(2); break; }; (...) # -------------------------------------------------- # Anonymous Call Rejection Section # -------------------------------------------------- if (isflagset(24) && (method=="INVITE") && search("^(f|F)rom:.*(a|A)nonymous")) { route(8); break; }; # -------------------------------------------------- # URI Compare Section # -------------------------------------------------- # Here we compare the "from" and "to" to see if the caller is dialing # their own extension. If so then we route to voicemail if needed if (method=="INVITE") { avp_write("$from", "i:34"); if (avp_check("i:34", "eq/$ruri/i")) { if (isflagset(31)) { route(5); break; } else { sl_send_reply("486", "Busy"); break; }; }; }; (...) # -------------------------------------------------- # Do Not Disturb Section # -------------------------------------------------- if (avp_db_load("$ruri/username", "s:donotdisturb")) { if (avp_check("s:donotdisturb", "eq/y/i")) { route(5); break; }; }; # -------------------------------------------------- # Call Routing Section # -------------------------------------------------- if (!lookup("location")){ # if flag 31 (ie voicemail) is set and we made it here this means # the user's phone is not registered anywhere. We'll forward to # voicemail after this block because we need to check the call # forward settings first if (isflagset(31)) { # flag 19 means the user has voicemail but is not online # so we need to remember to send to voicemail if call # forwarding is not enabled setflag(19); }; if (method=="INVITE") { if( !uri=~&quot;(a)fccn.pt&quot; ) { if (!method=="REGISTER") record_route(); t_relay(); break; } if (does_uri_exist()) { # o utilizador foi encontrado mas esta offline sl_send_reply("480", "Temporarily Unavailable"); break; }; sl_send_reply("404", "Utilizador nao Encontrado"); break; }; }; # -------------------------------------------------- # Call Forwarding Section # -------------------------------------------------- if (method=="INVITE") { # only load the forward no answer option if voice mail is not enabled if (!isflagset(31)) { if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { route(1); if (!avp_check("$calltype", "eq/-/i")) { if (avp_check("$calltype", "eq/dom/i")) { avp_write("dom", "$fwd_no_answer_type"); } else if (avp_check("$calltype", "eq/int/i")) { avp_write("int", "$fwd_no_answer_type"); } else { avp_write("sip", "$fwd_no_answer_type"); } setflag(27); }; }; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { route(1); if (!avp_check("$calltype", "eq/-/i")) { if (avp_check("$calltype", "eq/dom/i")) { avp_write("dom", "$fwd_busy_type"); } else if (avp_check("$calltype", "eq/int/i")) { avp_write("int", "$fwd_busy_type"); } else { avp_write("sip", "$fwd_busy_type"); } setflag(26); }; }; avp_pushto("$ruri", "i:99"); }; if (isflagset(19)) { # send to voicemail route(5); } else { route(2); }; } #route route[1] { avp_write("-", "$calltype"); if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # test for domestic PSTN number if (isflagset(28)) { avp_write("dom", "$calltype"); }; } else if (uri=~"^sip:011[0-9]*@") { # test for international PSTN number if (isflagset(29)) { avp_write("int", "$calltype"); }; }; break; }; avp_write("sip", "$calltype"); } route[2] { # check for RFC1918 ip addresses # -- NATHELPER -- if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; #check if the NAT Flag is set if (isflagset(6)) { log(1, "LOG: flag 6 set => One of the sides is NATed -> force_rtp_proxy()\n"); if(!force_rtp_proxy()) { log(1, "LOG: force_rtp_prpoxy FAILED!\n"); } else { log(1, "LOG: force_rtp_proxy() succeeded!\n"); }; }; if (isflagset(25)) { replace("^From:(.*)>" , "From: \"Anonymous\" <sip:someone at anonymous.invalid>"); }; if (method=="INVITE" || method=="ACK") { force_rtp_proxy(); }; t_on_failure("1"); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } # encaminhamento para o Gateway PSTN route[3] { if (method=="INVITE") { if (!proxy_authorize("fccn.pt", "utilizador")) { proxy_challenge("fccn.pt", "0"); break; } } else if (!check_from()) { log(1, "Spoofed SIP call attempt"); sl_send_reply("403", "Use From=ID"); break; } else if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Please register to use our service"); break; }; # enable caller id blocking for PSTN calls if (isflagset(25)) { append_rpid_hf(); }; }; # SIP->PSTN calls get 45 seconds to timeout avp_write("i:45", "inv_timeout"); rewritehostport("10.0.0.135:1720"); if (method=="INVITE" || method=="ACK") { force_rtp_proxy(); }; if (isflagset(31)) { t_on_failure("1"); }; t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } (...) onreply_route[1] { # Not all 2xx messages have a content body so here we # make sure our Content-Length > 0 to avoid a parse error if (status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { # forward busy is flag 26 if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # test for domestic PSTN gateway if (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway route(3); # sl_send_reply("503", "Gateway PSTN Indisponivel-3"); #} else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway # route(6); # } else { # default to sip call route(2); }; break; }; }; # here we can have either voicemail __OR__ forward no answer if (isflagset(27) && t_check_status("408")) { # forward no answer is flag 27 if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { # test for domestic PSTN gateway route(3); #sl_send_reply("503", "Gateway PSTN Indisponivel"); #} else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { # test for international PSTN gateway # route(6); #} else { # default to sip call route(2); }; break; }; } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { avp_delete("$voicemail"); route(4); break; }; } onreply_route[2] { log(1, "LOG: entered onreply_route[2] now... \n"); if(isflagset(6)) { # && status =~"(183)|2[0-9][0-9]") { log(1, "LOG: TRansaction was sent to a NATed client --> fix_nated_contact() and force_rtp_proxy().\n"); if(!fix_nated_contact()) { log(1, "LOG: fix_nated_contact() FAILED!\n"); } else { log(1, "LOG: fix_nated_contact() succeded.\n"); }; if(!force_rtp_proxy()) { log(1, "LOG: force_rtp_proxy() FAILED!\n"); } else { log(1, "LOG: force_rtp_proxy() succeded.\n"); }; append_hf("P-hint: fixed NAT contact for response\r\n"); } else if (nat_uac_test("1")) { log(1, "LOG: Uncaught NAT. => fix_nated_contact()\n"); if(!fix_nated_contact()) { log(1, "LOG: fix_nated_contact() FAILED!\n"); } else { log(1, "LOG: fix_nated_contact() succeded.\n"); }; }; } Sorry the long post. Joao Pereira www.fccn.pt Paul Hazlett wrote: